TW201906394A - Method, apparatus and computer readable storage medium for call management - Google Patents

Method, apparatus and computer readable storage medium for call management Download PDF

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TW201906394A
TW201906394A TW106122019A TW106122019A TW201906394A TW 201906394 A TW201906394 A TW 201906394A TW 106122019 A TW106122019 A TW 106122019A TW 106122019 A TW106122019 A TW 106122019A TW 201906394 A TW201906394 A TW 201906394A
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endpoint
call
proxy
available
conference
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TWI652932B (en
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廖俊雄
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鴻海精密工業股份有限公司
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Abstract

A method, apparatus and computer readable storage medium for call management are provided. The method for call management comprises storing a conference table for recording information of a proxy endpoint that has been added to a conference call and having a number of DSP channels greater than or equal to 2. The method further comprises receiving a call request; identifying one endpoint of the call request; determining whether there is a number of available proxy endpoints in the conference table, where the proxy endpoint is an available proxy endpoint if the number of the DSP channels of the proxy endpoint is greater than zero. If there are multiple available proxy endpoints in the conference table, the priority of each available proxy endpoint is calculated and the call request is redirected to the endpoint with the highest priority available proxy endpoint between.

Description

用於呼叫管理的方法、裝置及計算機可讀存儲介質Method, device and computer readable storage medium for call management

本發明涉及通信領域,尤其涉及一種用於呼叫管理的方法、裝置及計算機可讀存儲介質。The present invention relates to the field of communications, and in particular, to a method, apparatus, and computer readable storage medium for call management.

現有的網路電話機經由閘道器與公共交換電話網路(Public Switched Telephone Network, PSTN)相連接,可以實現多方電話會議。然而受限於標準電話中繼線的頻寬或是電話會議系統人數上限的設定,無論是由與會者主動撥入電話會議系統、或是由會議系統主動撥出給與會者,皆有可能面臨同時間佔用電話線路數量超過上限的問題。Existing VoIP phones are connected to the Public Switched Telephone Network (PSTN) via a gateway to enable multi-party conference calls. However, limited by the bandwidth of the standard telephone trunk or the maximum number of conference call system, whether the participant actively dials into the conference call system or is actively dialed out to the conference by the conference system, it is possible to face the same time. The problem that the number of occupied telephone lines exceeds the upper limit.

有鑒於此,需提供一種用於呼叫管理的方法、裝置及計算機可讀存儲介質,可以在有效降低電話會議對外建立呼叫數量的同時,增加電話會議的與會者端點數量。In view of the above, there is a need for a method, apparatus, and computer readable storage medium for call management that can increase the number of participants in a conference call while effectively reducing the number of calls established by the conference call.

本發明提供一種用於呼叫管理的方法,該方法包含:存儲一會議表,用於記錄已加入電話會議,且DSP通道數大於或等於2的代理人端點的信息;接收到一呼叫請求;識別該呼叫請求關聯的一端點;判斷該會議表是否存在多個可用的代理人端點,其中,若代理人端點的DSP通道數大於0,則該代理人端點為可用的代理人端點;若該會議表中存在多個可用的代理人端點,則計算各個可用的代理人端點的優先級;以及將該呼叫請求重新導向至該端點與優先級最高的可用的代理人端點間。The present invention provides a method for call management, the method comprising: storing a conference table for recording information of an agent endpoint that has joined a conference call and having a number of DSP channels greater than or equal to 2; receiving a call request; Identifying an endpoint associated with the call request; determining whether there are multiple available proxy endpoints in the conference table, wherein if the number of DSP channels of the proxy endpoint is greater than 0, the proxy endpoint is an available proxy endpoint Point; if there are multiple available agent endpoints in the conference table, calculate the priority of each available agent endpoint; and redirect the call request to the endpoint with the highest priority available agent Between endpoints.

本發明還提供一種用於呼叫管理的裝置,該裝置包含:一處理器;以及一計算機可讀存儲介質,該計算機可讀存儲介質用以存儲至少一個計算機程序及一會議表,用於記錄已加入一電話會議且DSP通道數大於或等於2的代理人端點的信息;其中該計算機程式包含指令集且由該處理器所執行,並執行包含下列步驟:接收一呼叫請求;識別該呼叫請求關聯的一端點;判斷該會議表是否存在多個可用的代理人端點,其中,若代理人端點的DSP通道數大於0,則該代理人端點為可用的代理人端點;若該會議表中存在多個可用的代理人端點,則計算各個可用的代理人端點的優先級;以及將該呼叫請求重新導向至該端點與優先級最高的可用的代理人端點間。The present invention also provides an apparatus for call management, the apparatus comprising: a processor; and a computer readable storage medium for storing at least one computer program and a conference table for recording Information of an agent endpoint joining a conference call and having a number of DSP channels greater than or equal to 2; wherein the computer program includes and is executed by the processor and performs the steps of: receiving a call request; identifying the call request An associated endpoint; determining whether there are multiple available proxy endpoints in the conference table, wherein if the number of DSP channels of the proxy endpoint is greater than 0, the proxy endpoint is an available proxy endpoint; There are multiple available agent endpoints in the conference table, then the priority of each available agent endpoint is calculated; and the call request is redirected to between the endpoint and the highest priority available proxy endpoint.

本發明還提供一種計算機可讀存儲介質,該計算機可讀存儲介質上存儲有計算機程序,所述計算機程序被處理器執行時實現所述用於呼叫管理方法的步驟。The present invention also provides a computer readable storage medium having stored thereon a computer program that, when executed by a processor, implements the steps for the call management method.

請參閱圖1,所示為根據本發明一實施例進行電話會議的應用環境100的示意圖。上述應用環境100包含第一場域110及第二場域150。在本實施例中,上述第一場域110經由PSTN140與上述第二場域150通信連接。需注意的是,上述第一場域110與上述第二場域150可以是位於不同的地理空間,也可以是位於同一地理空間的不同網路拓墣架構上的分群。Referring to FIG. 1, a schematic diagram of an application environment 100 for conducting a conference call according to an embodiment of the present invention is shown. The application environment 100 includes a first field 110 and a second field 150. In this embodiment, the first field domain 110 is communicably connected to the second field 150 via the PSTN 140. It should be noted that the first field 110 and the second field 150 may be located in different geographic spaces, or may be grouped on different network topologies in the same geographic space.

上述第一場域100包含多個端點110A-110M、閘道器120及呼叫管理器130。上述多個端點110A-110M為網際協議通話技術(Voice over Internet Protocol, VOIP)的客戶端,為終端使用者提供VOIP的服務。上述多個端點110A-110M可以是類比電話機、網路電話機或軟體電話機等終端設備。當上述多個端點110A-110M為軟體電話機時,可以運行在智慧型手機、個人電腦或其它媒介的使用者終端設備上。上述閘道器120可以耦合上述多個端點110A-110M,負責把呼叫轉發到上述PSTN140,完成異種網路的電話撥入和撥出。在一實施例中,上述閘道器120可以經由類比或數位中繼線,例如T1或E1接口,耦合到上述PSTN140。上述呼叫管理器130耦合到上述閘道器120,用於管理上述多個端點110A-110M的配置、註冊、呼叫管理和其他電話功能管理。The first field field 100 described above includes a plurality of endpoints 110A-110M, a gateway 120, and a call manager 130. The plurality of endpoints 110A-110M are clients of the Voice over Internet Protocol (VOIP), and provide VOIP services for the terminal users. The plurality of endpoints 110A-110M may be terminal devices such as analog telephones, network telephones, or soft telephones. When the plurality of endpoints 110A-110M are software telephones, they can be run on user terminals of smart phones, personal computers or other media. The gateway 120 can be coupled to the plurality of endpoints 110A-110M to forward calls to the PSTN 140 to complete dial-in and dial-out of heterogeneous networks. In an embodiment, the gateway 120 can be coupled to the PSTN 140 via an analog or digital trunk, such as a T1 or E1 interface. The call manager 130 described above is coupled to the gateway 120 for managing configuration, registration, call management, and other telephony function management of the plurality of endpoints 110A-110M.

上述第二場域150包含會議系統160、呼叫管理器170及閘道器180。上述會議系統160為多方電話會議的客戶端,用於提供使用者創建一個多方電話會議或參加多方電話會議。上述呼叫管理器170用於管理一個多方電話會議的創建、刪除、多方電話會議時間管理以及多方電話會議與會者管理等。上述閘道器180耦合到上述PSTN140,並向上述會議系統160及上述呼叫管理器170提供電話及網際協議通話技術服務。The second field field 150 described above includes a conference system 160, a call manager 170, and a gateway 180. The conference system 160 is a client of a multi-party conference call for providing a user to create a multi-party conference call or to participate in a multi-party conference call. The above call manager 170 is used to manage the creation, deletion, multi-party conference time management, multi-party conference call participant management, and the like of a multi-party conference call. The gateway 180 is coupled to the PSTN 140 and provides telephone and internet protocol call technology services to the conference system 160 and the call manager 170.

在一實施例中,上述呼叫管理器130、170包含處理器及計算機可讀存儲介質(圖一中未示出)。上述處理器為上述呼叫管理器130、170的中央處理器,可以由一個或多個積體電路,例如單核心或多核心微處理器或微控制器組成,以控制上述呼叫管理器130、170的操作。上述計算機可讀存儲介質可以是任何形式的計算機可讀取的存儲介質,例如硬碟、快閃記憶體或任何其他非揮發性存儲媒體。上述存儲裝置可存儲用於操作上述呼叫管理器130、170的一個或多個計算機程序。上述計算機程序可以由上述處理器執行,並包含有實現本發明特徵的指令集或功能單元。In one embodiment, the call manager 130, 170 includes a processor and a computer readable storage medium (not shown in FIG. 1). The above processor is a central processor of the call manager 130, 170, and may be composed of one or more integrated circuits, such as a single core or multi-core microprocessor or a microcontroller, to control the call manager 130, 170. Operation. The computer readable storage medium described above can be any form of computer readable storage medium such as a hard drive, flash memory or any other non-volatile storage medium. The storage device described above can store one or more computer programs for operating the call manager 130, 170 described above. The above computer program can be executed by the above described processor and includes a set of instructions or functional units that implement the features of the present invention.

在一實施例中,上述呼叫管理器130、170可以是運行在一個或多個計算機裝置上的計算機程序,上述呼叫管理器130、170的計算機程序可以包含在任何類型的計算機可讀存儲介質。需注意的是,儘管在圖1中,上述呼叫管理器130、170表示為單獨可操作的模組,但在其它實施例中,上述呼叫管理器130、170可以是運行在上述閘道器120、180中的計算機程序。In one embodiment, the call manager 130, 170 can be a computer program running on one or more computer devices, and the computer program of the call manager 130, 170 can be included in any type of computer readable storage medium. It should be noted that although the call manager 130, 170 is shown as a separately operable module in FIG. 1, in other embodiments, the call manager 130, 170 may be operated in the gateway 120. , 180 computer programs.

上述多個端點110A-110M可以包含相應的電話處理器,所述相應的電話處理器可以是數位信號處理器(Digital Signal Processor ,DSP),並且包含一個或多個DSP通道。在一實施例中,上述多個端點110A-110M在開機時以自身的媒體存取控制位址(Media Access Control Address, MAC Address)和其它參數自動地向上述呼叫管理器130註冊。上述其它參數可包含端點的類型、品牌、型號、使用者名稱及密碼等。上述呼叫管理器130根據上述多個端點110A-110M傳送的品牌及型號,可以得知各端點的DSP通道數,並進一步根據其DSP通道數判斷該端點在電話會議中可以擔任的角色。若端點的DSP通道數等於1,則該端點在一電話會議中僅能為與會者;若端點的DSP通道數大於或等於2,則該端點在一電話會議中可以為與會者也可以為代理人。在本發明實施例中,上述呼叫管理器130可以將欲加入上述會議系統160的與會者的呼叫重新導向至同一場域中的已加入至上述會議系統160的代理人,由該代理人進行後續語音資料流混合處理,以減少上述閘道器120同時對外的呼叫數量。The plurality of endpoints 110A-110M may include respective telephone processors, which may be digital signal processors (DSPs) and include one or more DSP channels. In one embodiment, the plurality of endpoints 110A-110M are automatically registered with the call manager 130 with their own Media Access Control Address (MAC Address) and other parameters when powered on. The other parameters mentioned above may include the type, brand, model, user name and password of the endpoint. The call manager 130 can know the number of DSP channels of each endpoint according to the brand and model transmitted by the multiple endpoints 110A-110M, and further determine the role that the endpoint can play in the conference call according to the number of DSP channels. . If the number of DSP channels of the endpoint is equal to 1, the endpoint can only be an attendee in a conference call; if the number of DSP channels of the endpoint is greater than or equal to 2, the endpoint can be a participant in a conference call. Can also be an agent. In the embodiment of the present invention, the call manager 130 may redirect the call of the participant who wants to join the conference system 160 to the agent who has joined the conference system 160 in the same field, and the caller performs the follow-up. The voice data stream is mixed to reduce the number of calls to the gateway 120 at the same time.

在一實施例中,上述呼叫管理器130可以及時通過演算法從多個代理人中選擇特定端點做為一新進呼叫的代理人,諸如如下等式:In an embodiment, the call manager 130 can select a specific endpoint from among a plurality of agents as an agent of a new incoming call by an algorithm in time, such as the following equation:

Ppriority = a´RTTlocal + b´QoSproxy + g´Capproxy + w´RTTexternal (1)P priority = a ́RTT local + b ́QoS proxy + g ́Cap proxy + w ́RTT external (1)

其中,上述等式計算得出的Ppriority 係可以做為代理人的端點的優先級,該值越小,優先級越高。上述呼叫管理器130會將一電話會議中新進的與會者的呼叫重新導向至具有最高優先級的代理人。The P priority calculated by the above equation can be used as the priority of the agent's endpoint, and the smaller the value, the higher the priority. The call manager 130 described above redirects the call of a new participant in a conference call to the agent with the highest priority.

RTTlocal 係新進與會者與已經撥入電話會議的代理人間的延遲時間(Round-Trip Time, RTT) ,該值可以由以下兩種方式取得:一、當上述多個端點110A-110M開機時,會主動廣播ICMP回送請求封包,藉以得到與其它端點間的延遲時間,並在註冊至上述呼叫管理器130後,將與其它端點間的延遲時間傳送給上述呼叫管理器130;二、當任兩個端點間進行通話,可以經由即時控制協定(Real-Time Control Protocol, RTCP)取得延遲時間,該端點於通話結束後,主動將該延遲時間傳送給上述呼叫管理器130。The RTT local is the Round-Trip Time (RTT) between the new participant and the agent who has dialed into the conference call. The value can be obtained in the following two ways: 1. When the above multiple endpoints 110A-110M are powered on, The ICMP echo request packet is actively broadcasted to obtain a delay time with other endpoints, and after registration to the call manager 130, the delay time with other endpoints is transmitted to the call manager 130; When a call is made between any two endpoints, the delay time can be obtained via a Real-Time Control Protocol (RTCP), and the endpoint actively transmits the delay time to the call manager 130 after the call ends.

QoSproxy 係代理人本身與服務品質(Quality of Service, QoS)有關的設定值。該QoS設定值可以從代理人的服務類型(Type of Service, ToS)設定值或差異式服務代碼點(Differentiated Services Code Point, DSCP)的設定值計算取得。一般而言,DSCP設定值的範圍為0~63,該DSCP設定值在QoS機制中是用於指定封包在QoS調度中的優先級,值越大表示優先級越高。若代理人有DSCP的設定值,可以直接將該DSCP設定值取出,以63減去該DSP設定值即為該QoS設定值。若代理人僅有ToS設定值,則需先將該ToS設定值映射至一DSCP設定值,再進行運算取得QoS設定值。代理人的ToS設定值或DSCP設定值的來源有以下兩種途徑:一、當代理人註冊至上述呼叫管理器130後,由上述呼叫管理器130傳送給該代理人的初始配置;二、使用者自行在端點上進行設定,再由該端點主動將該設定值傳送給上述呼叫管理器130。The QoS proxy is the set value of the agent itself related to the Quality of Service (QoS). The QoS setting value can be calculated from the agent's Type of Service (ToS) setting value or the differential service code point (DSCP) setting value. Generally, the DSCP setting value ranges from 0 to 63. The DSCP setting value is used to specify the priority of the packet in the QoS scheduling in the QoS mechanism. A larger value indicates a higher priority. If the agent has the DSCP setting value, the DSCP setting value can be directly taken out, and the DSP setting value is subtracted from 63 to be the QoS setting value. If the agent has only the ToS set value, the ToS set value must be mapped to a DSCP set value, and then the operation is performed to obtain the QoS set value. The source of the ToS set value or the DSCP set value of the agent has the following two ways: 1. After the agent registers with the call manager 130, the initial configuration is transmitted by the call manager 130 to the agent; The setting is performed on the endpoint by itself, and the endpoint actively transmits the set value to the call manager 130.

Capproxy 係代理人目前剩餘的DSP通道數。該值的初始值為端點本身硬體所支援的DSP通道數,上述呼叫管理器130可以根據端點註冊時所傳送的品牌及型號等資訊得知該端點支援的DSP通道數。在一電話會議進行中,隨著越來越多的新進與會者之呼叫重新導向至特定的代理人,該代理人的Capproxy 值亦隨之遞減。Cap proxy is the number of remaining DSP channels currently available to the agent. The initial value of the value is the number of DSP channels supported by the hardware of the endpoint itself. The call manager 130 can know the number of DSP channels supported by the endpoint according to information such as the brand and model transmitted when the endpoint is registered. During a conference call, as more and more incoming participants' calls are redirected to a particular agent, the agent's Cap proxy value is also decremented.

RTTexternal 係代理人與上述閘道器120間的延遲時間。在一實施例中,上述第一場域110還可以包含多個對外的閘道器,由上述呼叫管理器130根據設定將不同端點的呼叫導向至不同的對外閘道器,而端點與閘道器間的延遲時間可以經由RTCP取得。RTT external is the delay between the agent and the gateway 120 described above. In an embodiment, the first field 110 may further include a plurality of external gateways, and the call manager 130 directs calls of different endpoints to different external gateways according to the setting, and the endpoints are The delay time between the gateways can be obtained via RTCP.

a、b、g及w係權重參數,上述第一場域110的管理者或網管人員可以經由上述呼叫管理器130自行設定該等權重參數的初始值,設定後還可以根據不同需求及環境異動再加以調整,但總和必須等於1。該等權重參數的設定值建議如下:一、如果同一個場域中,存在數個高階端點,亦即存在數個高DSP通道數的端點,建議可以將g值調高,讓大部分的語音資料流都集中到某幾個代理人端點;二、如果在鄰近場域中,都存在著高階代理人端點,建議可以將a值調高,以減少混合處理後的語音延遲;三、如果網路環境中,存在數個QoS設定值較高的端點,建議可以將b值調高,以提高混合處理後的語音品質;四、如果網路環境中,不同的端點會有不同的路由路徑到不同的對外閘道器,建議可以將w值調高,以提高混合處理後的語音品質。在一實施例中,上述呼叫管理器130可以針對不同的電話會議儲存不同的會議表於上述計算機可讀存儲介質,用以記錄已加入電話會議之代理人的RTTlocal 、QoSproxy 、Capproxy 及RTTexternal 等參數值。上述會議表還可以包含可用性欄位,用於標記該代理人的可用性。其中,若該代理人目前剩餘的DSP通道數大於0,代表該代理人往後在電話會議中的角色可以為代理人,故相應地在可用性欄位應標記為可用;若該代理人目前剩餘的DSP通道數等於0時,代表該代理人往後在電話會議中的角色僅能為與會者,故相應地在可用性欄位應標記為不可用。The a, b, g, and w system weight parameters, the manager or network administrator of the first field domain 110 may set the initial values of the weight parameters by using the call manager 130, and may also change according to different needs and environments. Adjust again, but the sum must be equal to 1. The set values of the weight parameters are suggested as follows: 1. If there are several high-order endpoints in the same field, that is, there are several endpoints with a high number of DSP channels, it is recommended to increase the g value, so that most The voice data stream is concentrated to a certain number of agent endpoints; second, if there are high-level agent endpoints in the adjacent field, it is recommended to increase the a value to reduce the voice delay after the mixing process; 3. If there are several endpoints with higher QoS settings in the network environment, it is recommended to increase the b value to improve the voice quality after the hybrid processing. Fourth, if the network environment, different endpoints will There are different routing paths to different external gateways. It is recommended to increase the w value to improve the speech quality after mixing. In an embodiment, the call manager 130 may store different conference tables for the different conference calls on the computer readable storage medium to record the RTT local , QoS proxy , and Cap proxy of the agent who has joined the conference call. RTT external and other parameter values. The above conference table may also contain an availability field for marking the availability of the agent. Wherein, if the agent currently has more DSP channels than 0, the role of the agent in the conference call may be an agent, so the availability field should be marked as available; if the agent is currently available When the number of DSP channels is equal to 0, the role of the agent in the conference call can only be a participant, so the availability field should be marked as unavailable accordingly.

請參閱圖2,所示為根據本發明一實施例進行主動式電話會議時的呼叫管理流程圖。在本實施例中,以上述端點110A及110B分別擁有1個DSP通道,以及上述端點110C及110D分別擁有2個DSP通道為例。由於上述端點110A及110B僅擁有一個DSP通道,故在一電話會議中,僅能為與會者;而上述端點110C及110D分別擁有2個DSP通道,故在一電話會議中,可以為與會者,也可以為代理人。圖2所示為上述端點110A、110B、110C及110D與上述會議系統160建立主動式電話會議的流程,主要步驟如下:Referring to FIG. 2, a flow chart of call management when an active conference call is performed according to an embodiment of the present invention is shown. In this embodiment, each of the endpoints 110A and 110B has one DSP channel, and the endpoints 110C and 110D respectively have two DSP channels as an example. Since the above-mentioned endpoints 110A and 110B only have one DSP channel, they can only be participants in a conference call; and the endpoints 110C and 110D respectively have two DSP channels, so in a conference call, it is possible to participate in the conference. It can also be an agent. FIG. 2 shows a flow of establishing an active conference call with the above-mentioned endpoints 110A, 110B, 110C, and 110D and the conference system 160. The main steps are as follows:

步驟201,上述端點110C撥打上述會議系統160的電話號碼,上述呼叫管理器130接收到上述端點110C的呼叫請求。上述呼叫管理器130由上述端點110C的DSP通道數判斷其角色可以為與會者,也可以為代理人。In step 201, the endpoint 110C dials the telephone number of the conference system 160, and the call manager 130 receives the call request of the endpoint 110C. The call manager 130 determines that the role of the endpoint 110C is the number of DSP channels, and the role may be an attendee or an agent.

步驟202,上述呼叫管理器130建立上述端點110C與上述會議系統160間的通信連接,並通知上述端點110C開啟多媒體連接埠,開始傳送與接收語音資料流。上述呼叫管理器130 還將上述端點110C的相關參數,例如RTTlocal 、QoSproxy 、Capproxy 及RTTexternal 等,記錄於會議表。其中,上述端點110C的DSP通道數原本為2,在其加入上述會議系統160的電話會議後,上述呼叫管理器130將上述端點110C目前剩餘的DSP通道數更新為1,並記錄於會議表中的Capproxy 參數欄位,且在可用性欄位標記為可用。Step 202: The call manager 130 establishes a communication connection between the endpoint 110C and the conference system 160, and notifies the endpoint 110C to open the multimedia interface to start transmitting and receiving voice data streams. The call manager 130 also records related parameters of the endpoint 110C, such as RTT local , QoS proxy , Cap proxy, and RTT external , in the conference table. The number of DSP channels of the endpoint 110C is originally 2. After the conference call of the conference system 160 is added, the call manager 130 updates the number of remaining DSP channels of the endpoint 110C to 1, and records the conference. The Cap proxy parameter field in the table is marked as available in the Availability field.

步驟203,上述端點110A撥打上述會議系統160的電話號碼,上述呼叫管理器130接收到上述端點110A的呼叫請求。上述呼叫管理器130由上述端點110A的DSP通道數判斷其僅能為與會者。Step 203, the endpoint 110A dials the telephone number of the conference system 160, and the call manager 130 receives the call request of the endpoint 110A. The call manager 130 described above is judged by the number of DSP channels of the above-mentioned endpoint 110A to be only an attendee.

步驟204,由於目前參與上述會議系統160的代理人僅有上述端點110C,在本實施例中,上述呼叫管理器130決定上述端點110A的代理人為上述端點110C。Step 204: Since the agent currently participating in the conference system 160 has only the foregoing endpoint 110C, in the embodiment, the call manager 130 determines that the agent of the endpoint 110A is the endpoint 110C.

步驟205,上述呼叫管理器130將上述端點110A的呼叫請求重新導向至上述端點110C,並建立上述端點110A與上述端點110C間的通信連接。Step 205, the call manager 130 redirects the call request of the endpoint 110A to the endpoint 110C, and establishes a communication connection between the endpoint 110A and the endpoint 110C.

步驟206,上述呼叫管理器130通知上述端點110C開啟多媒體連接埠,處理上述端點110A的語音資料流,並進行後續的混合處理。由於上述端點110C已接收上述端點110A的語音資料流,其目前剩餘DSP通道數遞減為0,故上述呼叫管理器130更新會議表中關於上述端點110C的目前剩餘DSP通道數為0,並於可用性欄位中標記為不可用。In step 206, the call manager 130 notifies the endpoint 110C to open the multimedia port, processes the voice data stream of the endpoint 110A, and performs subsequent mixing processing. Since the endpoint 110C has received the voice data stream of the endpoint 110A, the current number of remaining DSP channels is decremented to 0, so the call manager 130 updates the current remaining DSP channel number of the endpoint 110C in the conference table to 0. And marked as unavailable in the availability field.

步驟207,上述端點110B撥打上述會議系統160的電話號碼,上述呼叫管理器收到上述端點110B的呼叫請求後,由上述端點110B的DSP通道數判斷其僅能為與會者。Step 207, the endpoint 110B dials the telephone number of the conference system 160, and after receiving the call request of the endpoint 110B, the call manager determines that the endpoint can be a participant only by the number of DSP channels of the endpoint 110B.

步驟208,由於目前參與上述會議系統160的代理人僅有上述端點110C,且上述端點110C已無剩餘的DSP通道數,僅能為與會者,故建立上述端點110B與上述會議系統160間的通信連接。In step 208, since the agent currently participating in the conference system 160 has only the foregoing endpoint 110C, and the endpoint 110C has no remaining number of DSP channels, and can only be an attendee, the endpoint 110B and the conference system 160 are established. Communication connection between.

步驟209,上述呼叫管理器130通知上述端點110B開啟多媒體連接埠,開始傳送與接收語音資料流。此時,上述閘道器120撥出到上述PSTN140的呼叫數量為2,但共有 3個端點參與電話會議。In step 209, the call manager 130 notifies the endpoint 110B that the multimedia connection is enabled, and starts transmitting and receiving the voice data stream. At this time, the number of calls dialed by the gateway 120 to the PSTN 140 is 2, but a total of 3 endpoints participate in the conference call.

步驟210,上述端點110D撥打上述會議系統160的電話號碼,上述呼叫管理器130接收到上述端點110D的呼叫後,由上述端點110D的DSP通道數判斷其可以為與會者,也可以為代理人。Step 210: The endpoint 110D dials the telephone number of the conference system 160. After receiving the call of the endpoint 110D, the call manager 130 determines that the endpoint may be an attendee by the number of DSP channels of the endpoint 110D, or may be agent.

步驟211,上述呼叫管理器130建立上述端點110D與上述會議系統160間的通信連接,並通知上述端點110D開啟多媒體連接埠,開始傳送與接收語音資料流。上述呼叫管理器130 將上述端點110D的相關參數,例如RTTlocal 、QoSproxy 、Capproxy 及RTTexternal 等,記錄於會議表。其中,上述端點110D的DSP通道數原本為2,在其加入上述會議系統160的電話會議後,上述呼叫管理器130將上述端點110D目前剩餘的DSP通道數更新為1,並記錄於會議表中的Capproxy 參數欄位,且在可用性欄位標記為可用。Step 211, the call manager 130 establishes a communication connection between the endpoint 110D and the conference system 160, and notifies the endpoint 110D to enable the multimedia connection to start transmitting and receiving voice data streams. The call manager 130 records the relevant parameters of the endpoint 110D, such as RTT local , QoS proxy , Cap proxy, and RTT external , in the conference table. The number of DSP channels of the endpoint 110D is originally 2. After the conference call of the conference system 160 is added, the call manager 130 updates the current number of remaining DSP channels of the endpoint 110D to 1, and records the conference. The Cap proxy parameter field in the table is marked as available in the Availability field.

步驟212,因為有新進的代理人,故上述呼叫管理器130重新決定上述端點110B的代理人為上述端點110D。Step 212, because there is a new agent, the call manager 130 re-determines that the agent of the endpoint 110B is the endpoint 110D.

步驟213,上述呼叫管理器130將上述端點110B的呼叫重新導向至上述端點110D,並建立上述端點110B與上述端點110D間的通信連接。Step 213, the call manager 130 redirects the call of the endpoint 110B to the endpoint 110D, and establishes a communication connection between the endpoint 110B and the endpoint 110D.

步驟214,上述呼叫管理器130通知上述端點110D開啟多媒體連接埠,處理上述端點110B的語音資料流,並進行後續的混合處理。由於上述端點110D已接收上述端點110B的語音資料流,其目前剩餘DSP通道數遞減為0,故上述呼叫管理器130更新會議表中關於上述端點110D的目前剩餘DSP通道數為0,並於可用性欄位標記為不可用。此時,上述閘道器120撥出到上述PSTN140的呼叫數量仍為2,但共有4個端點參與電話會議。Step 214, the call manager 130 notifies the endpoint 110D to open the multimedia port, process the voice data stream of the endpoint 110B, and perform subsequent mixing processing. Since the endpoint 110D has received the voice data stream of the endpoint 110B, the current number of remaining DSP channels is decremented to 0, so the call manager 130 updates the current remaining DSP channel number of the endpoint 110D in the conference table to 0. And the availability field is marked as unavailable. At this time, the number of calls dialed by the gateway 120 to the PSTN 140 is still 2, but a total of 4 endpoints participate in the conference call.

若於電話會議進行中,有任一端點掉線,上述呼叫管理器130將更新會議表,並且等待使用者重撥,再依前述步驟流程處理該端點的呼叫請求。If any endpoint is dropped during the conference call, the call manager 130 will update the conference table and wait for the user to redial, and then process the call request of the endpoint according to the foregoing procedure.

請參閱圖3,所示為根據本發明一實施例進行被動式電話會議時的呼叫管理流程圖。在本實施例中,以上述端點110A及110B分別擁有1個DSP通道,以及上述端點110C及110D分別擁有2個DSP通道為例。由於上述端點110A及110B僅擁有一個DSP通道,故在一電話會議中,僅能為與會者;而上述端點110C及110D分別擁有2個DSP通道,故在一電話會議中,可以為與會者,也可以為代理人。圖3所示為上述端點110A、110B、110C及110D與上述會議系統160建立被動式電話會議的流程,主要步驟如下:Referring to FIG. 3, a flow chart of call management when a passive conference call is performed according to an embodiment of the present invention is shown. In this embodiment, each of the endpoints 110A and 110B has one DSP channel, and the endpoints 110C and 110D respectively have two DSP channels as an example. Since the above-mentioned endpoints 110A and 110B only have one DSP channel, they can only be participants in a conference call; and the endpoints 110C and 110D respectively have two DSP channels, so in a conference call, it is possible to participate in the conference. It can also be an agent. FIG. 3 shows a flow of establishing a passive conference call between the endpoints 110A, 110B, 110C, and 110D and the conference system 160. The main steps are as follows:

步驟301,上述呼叫管理器130接收到由上述會議系統160傳送的呼叫請求,被叫方為上述端點110C。上述呼叫管理器130將該請求傳送到上述端點110C後,接收到來自上述端點110C的確認回應,並由上述端點110C的DSP通道數判斷其可以為與會者,也可以為代理人。Step 301, the call manager 130 receives the call request transmitted by the conference system 160, and the called party is the endpoint 110C. After the call manager 130 transmits the request to the endpoint 110C, it receives an acknowledgment response from the endpoint 110C, and the number of DSP channels of the endpoint 110C determines that it can be an attendee or an agent.

步驟302,上述呼叫管理器130建立上述端點110C與上述會議系統160間的通信連接,並通知上述端點110C開啟多媒體連接埠,開始傳送與接收語音資料流。上述呼叫管理器130 還將上述端點110C的相關參數,例如RTTlocal 、QoSproxy 、Capproxy 及RTTexternal 等,記錄於會議表。其中,上述端點110C的DSP通道數原本為2,在其加入上述會議系統160的電話會議後,上述呼叫管理器130將上述端點110C目前剩餘的DSP通道數更新為1,並記錄於會議表中的Capproxy 參數欄位,且於可用性欄位標記為可用。Step 302: The call manager 130 establishes a communication connection between the endpoint 110C and the conference system 160, and notifies the endpoint 110C to enable the multimedia connection to start transmitting and receiving voice data streams. The call manager 130 also records related parameters of the endpoint 110C, such as RTT local , QoS proxy , Cap proxy, and RTT external , in the conference table. The number of DSP channels of the endpoint 110C is originally 2. After the conference call of the conference system 160 is added, the call manager 130 updates the number of remaining DSP channels of the endpoint 110C to 1, and records the conference. The Cap proxy parameter field in the table is marked as available in the Availability field.

步驟303,上述呼叫管理器130接收到由上述會議系統160傳送的呼叫請求,被叫方為上述端點110D。上述呼叫管理器130將該請求傳送到上述端點110D後,接收到來自上述端點110D的確認回應,並由上述端點110D的DSP通道數判斷其可以為與會者,也可以為代理人。Step 303, the call manager 130 receives the call request transmitted by the conference system 160, and the called party is the endpoint 110D. After the call manager 130 transmits the request to the endpoint 110D, it receives an acknowledgment response from the endpoint 110D, and the number of DSP channels of the endpoint 110D determines that it can be an attendee or an agent.

步驟304,上述呼叫管理器130建立上述端點110D與上述會議系統160間的通信連接,並通知上述端點110D開啟多媒體連接埠,開始傳送與接收語音資料流。上述呼叫管理器130 還將上述端點110D的相關參數,例如RTTlocal 、QoSproxy 、Capproxy 及RTTexternal 等,記錄於會議表。其中,上述端點110D的DSP通道數原本為2,在其加入上述會議系統160的電話會議後,上述呼叫管理器130將上述端點110D目前剩餘的DSP通道數更新為1,並記錄於會議表中的Capproxy 參數欄位,且於可用性欄位標記為可用。此時,上述PSTN140撥入到上述閘道器120的呼叫數量為2,共有 2個端點參與電話會議。In step 304, the call manager 130 establishes a communication connection between the endpoint 110D and the conference system 160, and notifies the endpoint 110D to enable the multimedia connection to start transmitting and receiving voice data streams. The call manager 130 also records related parameters of the endpoint 110D, such as RTT local , QoS proxy , Cap proxy, and RTT external , in the conference table. The number of DSP channels of the endpoint 110D is originally 2. After the conference call of the conference system 160 is added, the call manager 130 updates the current number of remaining DSP channels of the endpoint 110D to 1, and records the conference. The Cap proxy parameter field in the table is marked as available in the Availability field. At this time, the number of calls dialed into the gateway 120 by the PSTN 140 is 2, and a total of two endpoints participate in the conference call.

步驟305,上述呼叫管理器130接收到由上述會議系統160傳送的呼叫請求,被叫方為上述端點110A。上述呼叫管理器130將該請求傳送到上述端點110A後,接收到來自上述端點110A的確認回應,並由上述端點110A的DSP通道數判斷其僅能為與會者。Step 305, the call manager 130 receives the call request transmitted by the conference system 160, and the called party is the endpoint 110A. After the call manager 130 transmits the request to the endpoint 110A, it receives an acknowledgment response from the endpoint 110A and determines that it can only be a participant by the number of DSP channels of the endpoint 110A.

步驟306,由於已參加電話會議的代理人有兩個端點,且兩個端點在會議表中的可用性欄位皆標記可用,故上述呼叫管理器130將計算等式(1)得到各代理人的Ppriority 值,並根據Ppriority 值決定上述端點110A 的代理人。在本實施例中,以上述端點110C及上述端點110D的QoSproxy 設定值相同,而上述端點C的RTTlocal 值及RTTexternal 值皆小於上述端點110D的RTTlocal 值及RTTexternal 值為例。上述呼叫管理器130查找會議表,取得計算等式(1)所需的相關參數值,並計算得到上述端點110C的Ppriority 值小於上述端點D的Ppriority 值,亦即上述端點110C的優先級高於上述端點110D的優先級。上述呼叫管理器130根據各代理人的優先級決定上述端點110A的代理人為上述端點110C。Step 306, since the agent who has participated in the conference call has two endpoints, and the availability fields of both endpoints in the conference table are marked, the call manager 130 calculates the equation (1) to obtain the agents. The P priority value of the person, and the agent of the above endpoint 110A is determined according to the P priority value. In this embodiment, the QoS proxy setting values of the endpoint 110C and the endpoint 110D are the same, and the RTT local value and the RTT external value of the endpoint C are both smaller than the RTT local value and the RTT external value of the endpoint 110D. For example. Said call session lookup table manager 130 acquires calculation equation (1) related to the desired parameter values, and the calculated value of the end point P priority values P priority 110C is smaller than the end D, i.e. above 110C endpoint The priority is higher than the priority of the above endpoint 110D. The call manager 130 determines that the agent of the endpoint 110A is the endpoint 110C according to the priority of each agent.

步驟307,上述呼叫管理器130將呼叫重新導向至上述端點110C,並建立上述端點110A與上述端點110C間的通信連接。In step 307, the call manager 130 redirects the call to the endpoint 110C and establishes a communication connection between the endpoint 110A and the endpoint 110C.

步驟308,上述呼叫管理器130通知上述端點110C開啟多媒體連接埠,處理上述端點110A的語音資料流,並進行後續的混合處理。由於上述端點110C已接收上述端點110A的語音資料流,其目前剩餘DSP通道數由1遞減為0,故上述呼叫管理器130將上述端點110C的目前剩餘DSP通道數更新為0,並記錄於會議表中的Capproxy 參數欄位,且於可用性欄位標記為不可用。Step 308, the call manager 130 notifies the endpoint 110C to open the multimedia port, process the voice data stream of the endpoint 110A, and perform subsequent mixing processing. Since the endpoint 110C has received the voice data stream of the endpoint 110A, the current number of remaining DSP channels is decremented from 1 to 0, so the call manager 130 updates the current remaining DSP channel number of the endpoint 110C to 0, and The Cap proxy parameter field recorded in the conference table and marked as unavailable in the availability field.

步驟309,上述呼叫管理器130接收到由上述會議系統160傳送的呼叫請求,被叫方為上述端點110B。上述呼叫管理器130將該請求傳送到上述端點110B後,接收到來自上述端點110B的確認回應,並由上述端點110B的DSP通道數判斷其僅能為與會者。Step 309, the call manager 130 receives the call request transmitted by the conference system 160, and the called party is the endpoint 110B. After the call manager 130 transmits the request to the endpoint 110B, it receives an acknowledgment response from the endpoint 110B and determines that it can only be a participant by the number of DSP channels of the endpoint 110B.

步驟310,由於會議表中,上述端點110C已標記為不可用,故可用的代理人僅有上述端點110D。上述呼叫管理器130查找會議表後,決定上述端點110B的代理人為上述端點110D。In step 310, since the above endpoint 110C has been marked as unavailable in the conference table, the available agents have only the above endpoint 110D. After the call manager 130 searches the conference table, it determines that the agent of the endpoint 110B is the endpoint 110D.

步驟311,上述呼叫管理器130將呼叫重新導向至上述端點110D,並建立上述端點110B與上述端點110D間的通信連接。In step 311, the call manager 130 redirects the call to the endpoint 110D and establishes a communication connection between the endpoint 110B and the endpoint 110D.

步驟312,上述呼叫管理器130通知上述端點110D開啟多媒體連接埠,處理上述端點110B的語音資料流,並進行後續的混合處理。由於上述端點110D已接收上述端點110B的語音資料流,其目前剩餘DSP通道數由1遞減為0,故上述呼叫管理器130將上述端點110D目前剩餘的DSP通道數更新為0,並記錄於會議表中的Capproxy 參數欄位,且於可用性欄位標記為不可用。此時,上述PSTN140撥入到上述閘道器120的呼叫數量仍為2,但共有 4個端點參與電話會議。In step 312, the call manager 130 notifies the endpoint 110D to open the multimedia port, processes the voice data stream of the endpoint 110B, and performs subsequent mixing processing. Since the endpoint 110D has received the voice data stream of the endpoint 110B, the current number of remaining DSP channels is decremented from 1 to 0, so the call manager 130 updates the current remaining DSP channel number of the endpoint 110D to 0, and The Cap proxy parameter field recorded in the conference table and marked as unavailable in the availability field. At this time, the number of calls dialed into the gateway 120 by the PSTN 140 is still 2, but a total of 4 endpoints participate in the conference call.

被動式電話會議進行中,若有任一端點掉線,將採與主動式電話會議相同的呼叫管理流程。In the middle of a passive conference call, if any endpoint is dropped, the same call management process as the active conference call will be adopted.

請參閱圖4,所示為根據本發明一實施例之呼叫管理方法流程400的示意圖。上述流程400應用於上述呼叫管理器130中,主要包含以下步驟:Referring to FIG. 4, a schematic diagram of a call management method flow 400 in accordance with an embodiment of the present invention is shown. The above process 400 is applied to the call manager 130 described above, and mainly includes the following steps:

步驟S410,接收新的呼叫請求。上述呼叫管理器130收到新的呼叫請求後,還進行以下步驟:若為撥出的呼叫請求,經由被叫方的電話號碼確認是否為同一電話會議的呼叫請求;若為撥入的呼叫請求,則經由主叫方的電話號碼確認是否為同一電話會議的呼叫請求。若上述電話會議為新建立,上述呼叫管理器130亦新建一會議表用於管理已加入該電話會議的代理人。Step S410, receiving a new call request. After receiving the new call request, the call manager 130 further performs the following steps: if it is an outgoing call request, confirm whether it is a call request of the same conference call via the called party's telephone number; if it is an incoming call request Then, it is confirmed by the calling party's telephone number whether it is a call request of the same conference call. If the conference call is newly established, the call manager 130 also creates a new conference table for managing the agents who have joined the conference call.

步驟S420,判斷呼叫請求相關的端點是否僅能為上述電話會議的與會者。其中,具體判斷步驟如下:若為撥出的呼叫請求,根據主叫方端點的DSP通道數進行判斷;若為撥入的呼叫請求,則根據被叫方端點的DSP通道數進行判斷。若上述端點的DSP通道數等於1,則該端點在該電話會議中僅能為與會者;若上述端點的DSP通道數大於或等於2,則該端點在該電話會議中可以為與會者也可以為代理人。若上述端點僅能為上述電話會議的與會者,則上述流程400繼續執行步驟S430;否則,跳轉至步驟S460。In step S420, it is determined whether the endpoint related to the call request can only be the participant of the above conference call. The specific judging step is as follows: if it is a dialed call request, it is judged according to the number of DSP channels of the calling party endpoint; if it is an incoming call request, it is judged according to the number of DSP channels of the called party endpoint. If the number of DSP channels of the foregoing endpoint is equal to 1, the endpoint can only be an attendee in the conference call; if the number of DSP channels of the endpoint is greater than or equal to 2, the endpoint may be in the conference call. Participants can also be agents. If the foregoing endpoint can only be the participant of the conference call, the process 400 continues to perform step S430; otherwise, the process branches to step S460.

步驟S430,上述呼叫管理器130根據會議表判斷是否已經有代理人參加電話會議。若已有代理人參加電話會議,則於步驟S432進一步判斷是否有多個可用的代理人。若有多個可用的代理人,則於步驟S440依據等式(1)計算各代理人的優先級,以優先級最高的代理人作為該上述端點的代理人,將該呼叫重新導向至該代理人,接著執行步驟S460的建立通話。若無多個可用的代理人,則於步驟S434進一步判斷是否有一個可用的代理人。若有一個可用的代理人,則於步驟S450中將該呼叫重新導向至該代理人,接著執行步驟S460的建立通話;若無任何可用的代理人,則直接執行步驟S460的建立通話。In step S430, the call manager 130 determines, according to the conference table, whether an agent has participated in the conference call. If an agent has participated in the conference call, it is further determined in step S432 whether there are a plurality of available agents. If there are multiple agents available, the priority of each agent is calculated according to equation (1) in step S440, and the agent with the highest priority is used as the agent of the terminal, and the call is redirected to the The agent then performs the setup call of step S460. If there are no more available agents, it is further determined in step S434 whether there is an available agent. If there is an available agent, the call is redirected to the agent in step S450, and then the setup call of step S460 is performed; if there is no available agent, the setup call of step S460 is directly performed.

通話建立完成後,步驟S470中,上述呼叫管理器130進一步判斷是否須更新會議表中與代理人有關的相關資訊,若新進的呼叫已重新導向至代理人或上述端點可以為代理人,則執行步驟S480,更新會議表中的代理人資訊,尤其是記錄目前剩餘DSP通道數的Capproxy 參數欄位以及表示可用性的可用性欄位;否則,結束上述流程400。After the call setup is completed, in step S470, the call manager 130 further determines whether the relevant information related to the agent in the conference table needs to be updated, and if the new call has been redirected to the agent or the endpoint may be an agent, then Step S480 is executed to update the agent information in the conference table, in particular, the Cap proxy parameter field for recording the current remaining DSP channel number and the availability field indicating availability; otherwise, the above process 400 is ended.

在另一實施中,於上述步驟S420中,若判斷上述端點可以為代理人,在執行完步驟S460至步驟S480後,上述呼叫管理器130還進一步判斷是否有任何的與會者端點未被重新導向至代理人端點,若有未被重新導向的與會者端點,則上述呼叫管理器130將執行上述流程400的步驟S430至步驟S480,為該未被重新導向的與會者端點選定重新導向的代理人。In another implementation, in the foregoing step S420, if it is determined that the endpoint may be an agent, after performing steps S460 to S480, the call manager 130 further determines whether any participant endpoint is not Redirecting to the agent endpoint, if there is an unredirected participant endpoint, the call manager 130 will perform steps S430 through S480 of the above-described process 400 to select the unredirected participant endpoint. Redirected agent.

在另一實施中,為了在盡量不增加同一場域中對外撥出或對內撥入的呼叫數量的情況下,盡量增加參與同一電話會議的端點數量,上述呼叫管理器130也可以在上述流程400的步驟S410之後,直接執行上述步驟S430,使得同一電話會議中的代理人端點不僅可以處理僅能為與會者端點的語音資料流,也可以處理同為代理人端點的語音資料流。In another implementation, in order to increase the number of endpoints participating in the same conference call as much as possible without increasing the number of calls dialed in or out of the same field, the call manager 130 may also be in the above manner. After step S410 of the process 400, the above step S430 is directly performed, so that the agent endpoint in the same conference call can process not only the voice data stream that can only be the participant endpoint but also the voice data of the same agent endpoint. flow.

總結來說,上述呼叫管理器130的呼叫管理方法,因為等式(1)中的參數已考量網路環境,故可以在有效降低電話會議所佔用上述閘道器120對外電話線路數量的同時,保有一定的會議通話品質。上述呼叫管理方法還可以完全支援主動式及被動式電話會議系統,且對於端點的使用者而言,無需改變其使用習慣就能有更好的使用者體驗。In summary, the call management method of the call manager 130 described above can effectively reduce the number of external telephone lines of the gateway 120 occupied by the conference call, because the parameters in the equation (1) have taken into account the network environment. Maintain a certain conference call quality. The above call management method can fully support the active and passive teleconferencing systems, and the end user can have a better user experience without changing their usage habits.

綜上所述,本發明符合發明專利要件,爰依法提出專利申請。惟,以上所述者僅為本發明之較佳實施例,舉凡熟悉本案技藝之人士,在爰依本案發明精神所作之等效修飾或變化,皆應包含於以下之申請專利範圍內。In summary, the present invention complies with the requirements of the invention patent and submits a patent application according to law. The above description is only the preferred embodiment of the present invention, and equivalent modifications or variations made by those skilled in the art of the present invention should be included in the following claims.

100‧‧‧應用環境 100‧‧‧Application environment

110‧‧‧第一場域 110‧‧‧First field

120,180‧‧‧閘道器 120,180‧‧‧ gateway

130,170‧‧‧呼叫管理器 130,170‧‧‧Call Manager

110A,110B-110M‧‧‧端點 110A, 110B-110M‧‧‧ endpoint

140‧‧‧PSTN 140‧‧‧PSTN

150‧‧‧第二場域 150‧‧‧ second field

160‧‧‧會議系統 160‧‧‧Conference system

圖1為用於呼叫管理之應用環境一實施例的示意圖。 圖2為主動式會議時呼叫管理一實施例的流程圖。 圖3為被動式會議時呼叫管理一實施例的流程圖。 圖4為呼叫管理方法一實施例的流程圖。1 is a schematic diagram of an embodiment of an application environment for call management. 2 is a flow chart of an embodiment of call management during active conferences. 3 is a flow chart of an embodiment of call management during passive conferences. 4 is a flow chart of an embodiment of a call management method.

藉由以下對具體實施例詳細的描述結合附圖,將可輕易的瞭解上述內容及此項發明之諸多優點。The above and many advantages of the invention will be readily apparent from the following detailed description of the preferred embodiments.

Claims (13)

一種用於呼叫管理的方法,該方法包含: 存儲一會議表,用於記錄已加入電話會議,且DSP通道數大於或等於2的代理人端點的信息; 接收一呼叫請求; 識別該呼叫請求關聯的一端點; 判斷該會議表是否存在多個可用的代理人端點,其中,若代理人端點的DSP通道數大於0,則該代理人端點為可用的代理人端點; 若該會議表中存在多個可用的代理人端點,則計算各個可用的代理人端點的優先級;以及 將該呼叫請求重新導向至該端點與優先級最高的可用的代理人端點間。A method for call management, the method comprising: storing a conference table for recording information of an agent endpoint that has joined a conference call and having a number of DSP channels greater than or equal to 2; receiving a call request; identifying the call request An associated endpoint; determining whether there are multiple available proxy endpoints in the conference table, wherein if the number of DSP channels of the proxy endpoint is greater than 0, the proxy endpoint is an available proxy endpoint; There are multiple available agent endpoints in the conference table, then the priority of each available agent endpoint is calculated; and the call request is redirected to between the endpoint and the highest priority available proxy endpoint. 如申請專利範圍第1項所述用於呼叫管理的方法,其中所述識別該呼叫請求關聯的一端點包括若該呼叫請求為撥出呼叫,則該端點為主叫方;若該呼叫請求為撥入呼叫,則該端點為被叫方。The method for call management according to claim 1, wherein the identifying an endpoint associated with the call request comprises: if the call request is an outgoing call, the endpoint is a calling party; if the call request For an incoming call, the endpoint is the called party. 如申請專利範圍第2項所述用於呼叫管理的方法,其中所述各個可用的代理人端點的優先級計算方法為: a´RTTlocal + b´QoSproxy + g´Capproxy + w´RTTexternal , 其中,a、b、g及w係權重參數;RTTlocal 係該端點與該可用的代理人端點間的延遲時間;QoSproxy 係該可用的代理人端點的QoS設定值;Capproxy 係該可用的代理人端點目前剩餘的DSP通道數;以及RTTexternal 係該可用的代理人端點對外傳送封包的延遲時間。A method for call management as described in claim 2, wherein the priority calculation method of each available agent endpoint is: a ́RTT local + b ́QoS proxy + g ́Cap proxy + w RTT external , where a, b, g, and w are weight parameters; RTT local is the delay time between the endpoint and the available proxy endpoint; QoS proxy is the QoS setting of the available proxy endpoint; Cap proxy is the current number of DSP channels remaining for the available proxy endpoint; and RTT external is the delay time for the available proxy endpoint to forward the packet. 如申請專利範圍第3項所述用於呼叫管理的方法,其中a、b、g及w係權重參數,總和為1。A method for call management as described in claim 3, wherein a, b, g, and w are weighting parameters, and the sum is 1. 如申請專利範圍第3項所述用於呼叫管理的方法,其中a´RTTlocal + b´QoSproxy + g´Capproxy + w´RTTexternal 的值越小,優先級越高。The method for call management as described in claim 3, wherein the smaller the value of a ́RTT local + b ́QoS proxy + g ́Cap proxy + w ́RTT external , the higher the priority. 如申請專利範圍第1項所述用於呼叫管理的方法,還包括,將該呼叫請求重新導向至該端點與優先級最高的可用的代理人端點間之後,根據該端點與該可用的代理人端點目前剩餘的DSP通道數,更新該會議表。The method for call management according to claim 1, further comprising, after redirecting the call request to the endpoint and the highest priority available proxy endpoint, according to the endpoint and the available The agent endpoint is currently updating the conference table by the number of remaining DSP channels. 一種用於呼叫管理的裝置,該裝置包含: 一處理器;以及 一計算機可讀存儲介質,該計算機可讀存儲介質用以存儲至少一個計算機程序及一會議表,用於記錄已加入一電話會議且DSP通道數大於或等於2的代理人端點的信息; 其中該計算機程式包含指令集且由該處理器所執行,並執行包含下列步驟: 接收一呼叫請求; 識別該呼叫請求關聯的一端點; 判斷該會議表是否存在多個可用的代理人端點,其中,若代理人端點的DSP通道數大於0,則該代理人端點為可用的代理人端點; 若該會議表中存在多個可用的代理人端點,則計算各個可用的代理人端點的優先級;以及 將該呼叫請求重新導向至該端點與優先級最高的可用的代理人端點間。An apparatus for call management, the apparatus comprising: a processor; and a computer readable storage medium for storing at least one computer program and a conference table for recording that a conference call has been joined And the information of the agent endpoint of the DSP channel number greater than or equal to 2; wherein the computer program includes the instruction set and is executed by the processor, and the executing includes the following steps: receiving a call request; identifying an endpoint associated with the call request Determining whether there are multiple available proxy endpoints in the conference table, wherein if the number of DSP channels of the proxy endpoint is greater than 0, the proxy endpoint is an available proxy endpoint; if the conference table exists A plurality of available agent endpoints, the priority of each available agent endpoint is calculated; and the call request is redirected to between the endpoint and the highest priority available agent endpoint. 如申請專利範圍第7項所述用於呼叫管理的裝置,其中所述識別該呼叫請求關聯的一端點包括若該呼叫請求為撥出呼叫,則該端點為主叫方;若該呼叫請求為撥入呼叫,則該端點為被叫方。The apparatus for call management of claim 7, wherein the identifying an endpoint associated with the call request comprises: if the call request is an outgoing call, the endpoint is a calling party; if the call request For an incoming call, the endpoint is the called party. 如申請專利範圍第8項所述用於呼叫管理的裝置,其中所述各個可用的代理人端點的優先級計算方法為: a´RTTlocal + b´QoSproxy + g´Capproxy + w´RTTexternal , 其中,a、b、g及w係權重參數;RTTlocal 係該端點與該可用的代理人端點間的延遲時間;QoSproxy 係該可用的代理人端點的QoS設定值;Capproxy 係該可用的代理人端點目前剩餘的DSP通道數;以及RTTexternal 係該可用的代理人端點對外傳送封包的延遲時間。The apparatus for call management according to claim 8, wherein the priority calculation method of each available agent endpoint is: a ́RTT local + b ́QoS proxy + g ́Cap proxy + w RTT external , where a, b, g, and w are weight parameters; RTT local is the delay time between the endpoint and the available proxy endpoint; QoS proxy is the QoS setting of the available proxy endpoint; Cap proxy is the current number of DSP channels remaining for the available proxy endpoint; and RTT external is the delay time for the available proxy endpoint to forward the packet. 如申請專利範圍第9項所述用於呼叫管理的裝置,其中a、b、g及w係權重參數,總和為1。The device for call management according to claim 9, wherein a, b, g, and w are weighting parameters, and the sum is 1. 如申請專利範圍第9項所述用於呼叫管理的裝置,其中a´RTTlocal + b´QoSproxy + g´Capproxy + w´RTTexternal 的值越小,優先級越高。The apparatus for call management according to claim 9 of the patent application, wherein the smaller the value of a ́RTT local + b ́QoS proxy + g ́Cap proxy + w ́RTT external , the higher the priority. 如申請專利範圍第7項所述用於呼叫管理的裝置,其中該指令集還執行下列步驟:將該呼叫請求重新導向至該端點與優先級最高的可用的代理人端點間之後,根據該端點與該可用的代理人端點目前剩餘的DSP通道數,更新該會議表。The apparatus for call management according to claim 7, wherein the instruction set further performs the following steps: after redirecting the call request to the endpoint and the highest priority available proxy endpoint, according to The conference table is updated with the number of DSP channels currently remaining by the endpoint and the available agent endpoint. 一種計算機可讀存儲介質,該計算機可讀存儲介質上存儲有計算機程序,該算機程序被處理器執行時實現如申請專利範圍第1項至第6項中任一項所述的用於呼叫管理方法的步驟。A computer readable storage medium having stored thereon a computer program, the computer program being executed by a processor for implementing a call as claimed in any one of claims 1 to 6 The steps of the management method.
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